The present invention relates generally to telecommunications, and more specifically, to a method and system of establishing and managing telecommunications over telecommunication networks by providing and enforcing a warranty for communications failing to meet parameters agreed to by the entities involved in the communication.
For many decades, telecommunication networks were designed to carry human voices and some data signals, such as Morse Code, in audio bands. However, in the last few decades, telecommunications have been moving into higher bandwidths, using digital signals in order to increase the capacity of physical infrastructures and reducing the cost of supplying telecommunication services. While a copper wire pair in a traditional telephone system carried a single analog voice signal, dozens of voice signals can now be digitized, multiplexed, and transmitted at higher frequencies over the same copper wire pair.
Telecommunication Service Providers now use various transmission means including analog, digital and compressed digital methods, over a variety of media, including hard wire, wireless, fiber optic and satellite transmission means. Data transmission methods and protocols now include Internet Protocol (IP), asynchronous transfer mode (ATM), frame relay, and digital telephony. The networks of these Service Providers are generally interconnected with those of others to form larger, heterogeneous networks.
Currently, two networks dominate telecommunications: conventional telephone networks (public switched telephone networks or PSTNs) with their almost universal physical infrastructure; and the Internet, which has grown tremendously over the last decade and continues to grow.
Telecommunication systems, such as those for telephony and the Internet, are composed of terminal equipment such as telephones or personal computers; an access network such as a telephony local loop or a radio link, switches or routers; and a backbone network such as the PSTN or an intercity data network. One design challenge is that the needs of users at the terminals are very varied, but the backbone networks must handle highly standardized loads in order to operate reliably and efficiently.
FIG. 1 is an example of a prior art telephony system 10. System 10 includes a plurality of switches 12 controlled by large computer programs in switch controllers 14. Switches 12 are interconnected with one another by trunks 16 which carry the actual communication signals and can consist of a variety of physical media, such as optical fiber and coaxial cables. Switch controllers 14 are also interconnected, generally by means of signaling lines 18 rather than over communication trunks 16.
Telephony systems 10 also generally include computing means to implement such features as conference calling 20, voice mail 22 and toll services 24. Telephony features, such as call forwarding, may be implemented by adding code to the programs running the switches 12 or by adding specialized hardware to the telephony system 10. The features available to particular users are defined in databases accessed by software on switch 12, and adding a new type of feature may involve changing these databases together with the software on each switch 12 that uses them, and may also involve purchasing and installing new types of hardware in the system.
FIG. 2 is an example of a prior art Internet communications system 30. The Internet 32 itself is represented by a number of routers 34 interconnected by an Internet backbone 36 network designed for high-speed transport of large amounts of data. User""s computers 38 may access the Internet in a number of manners including modulating and demodulating data over a telephone line using audio frequencies. Such dial up access requires a modem 40 and connection to the Public Switched Telephone Network 42, which in turn connects to the Internet 32 via a point of presence 44 including a complementary modem 40 and an access controller 46. Another manner of access is the use of broadband modems 50 which modulate and demodulate data onto high frequencies which pass over CATV networks 52, or the like, which are connected to the Internet via a controller 54.
Part of the access network in these systems is usually a set of computer systems 39 at the edge of the backbone network 36 which perform functions such as authentication of users and control of the load that they place on the backbone network 36. Communications between users"" computers 38 and the rest of the network 30 are standardized by means of defined communication protocols.
Communications over the Internet can be accomplished via various protocols and over a variety of physical transfer media. A protocol is a set of conventions or rules that govern transfer of data between hardware devices. The simplest protocols define only a hardware configuration while more complex protocols define timing, data formats, packet construction and interpretation, error detection and correction techniques and software structures.
The Internet is a connectionless network service, in that a single communication may be broken up into a multitude of data packets that follow different paths between the same source and destination. Traditional telephony, in contrast, reserves resources to establish a single dedicated path for a communication that all of the data in the communication follows.
The Internet employs the Internet Protocol (IP) and the key advantages of IP are that it allows a large network to function robustly and that it offers a standardized means by which applications software can use that network. While it offers a number of advantages, actual performance is based on performance levels which are not consistent or absolutely guaranteed and which can, at best, only be statistically estimated.
Networks for telephony and data transmission have developed separately, but the economic rationale for having distinct physical networks is disappearing and the technologies are converging. They appear to be converging on a model closer to that for data than that for telephony, partly because of the greater generality of data networks. The dominant data network is currently the Internet but there is a fundamental difference between these two networks. Conventional telephone systems generally take a xe2x80x9cfirst-come-first-servedxe2x80x9d approach when there is contention for network resources, denying services to subsequent callers if sufficient resources are not available and this process is known as xe2x80x9ccall admissionxe2x80x9d. The Internet however, is packet based and has traditionally offered xe2x80x9cbest effortxe2x80x9d service without making any attempt to prioritize traffic. That is, the Internet will accept all traffic, and the flow-through rate will vary with the demands the parties place on the resources available. This difference in operating philosophy makes it difficult to offer traditional services over a converging network.
As well, because the requirements for voice and data transmission can be quite different, it is difficult to optimize for provision of both on a common network. Voice communication, for example, produces a relatively steady stream of data at a relatively low data rate, and rapid delivery is more important than accuracy (i.e. a low end to end latency is more important than a small percentage of dropped packets). In contrast, data applications such as Web browsing or ftp file transfers generally produce bursts of data that are required to be delivered accurately, but for which an end to end latency of a second or two or more may be considered acceptable.
This problem is aggravated by the demand for new services such as video telephony, Internet games, video on demand, Internet audio, streaming audio or video, remote collaborative work or telemedicine, which require differing levels of quality and degrees of bandwidth. Clearly, the network must be able to allocate and control the quality and quantity of bandwidth in order to use its resources efficiently and to meet the needs of its users.
For example, telemedicine surgery in which a physician uses a remote manipulator to perform surgery could likely not be implemented with the existing Internet. This application would require very strict demands on both accuracy and timeliness together with a high bandwidth for video. The consequences of the network failing to perform as required would be very serious.
A contrasting example is multiplayer gaming, in which a number of players exchange small packets of information to update each other on their moves and present state. Given how such games are typically implemented, this can require low latencies, but bandwidth requirements are light and a fairly high rate of packet loss can be tolerated.
Existing networks are not designed to provide such diverse services and performance requirements.
While the Internet provides an efficient network for transporting data packets, it is not designed to provide end to end services with guaranteed performance levels. Typically, there is a static selection of services available to users, under predetermined terms and conditions. The performance level that a user may expect is offered on a xe2x80x9cbest effortxe2x80x9d basis and is not firmly guaranteed.
The Internet has attempted to provide guaranteed quality of service (QoS) by use of the resource reservation protocol (RSVP). RSVP is an extension to IP that permits specification of quality of service at a technical level, in terms of parameters such as data rates and latencies by reserving network resources to establish a xe2x80x98virtual connectionxe2x80x99 with the required QoS. It has had limited acceptance due to the complexity it adds to backbone networks and the need for their switching hardware to be updated, and it fails to include mechanisms to specify the costs associated with the QoS demands that it makes. More significantly, RSVP ensures quality of service by reserving resources, a strategy which lacks the efficiencies of the best-effort networks as it can result in the reserved resources being idle at various times.
Asynchronous Transfer Mode (ATM) networks use standard protocols for addressing packets of data (as does IP), setting up connections (as does TCP), and specifying QoS (as does RSVP). ATM networks have typically been deployed in the core of backbone networks because of the high speeds at which ATM equipment operates, but ATM capabilities have not been directly visible to end users (because of the dominance of IP as an applications standard and the high costs of ATM equipment). Because ATM routers are not directly accessible and because of the complexity of their mechanisms for describing QoS, these mechanisms have not been used by applications software. Further, reservation systems such as ATM or RSVP only deal with network capacity and can still fail to meet performance requirements due to equipment failures, etc. Also, as was the case with RSVP, these QoS mechanisms do not include methods by which to describe the costs associated with a QoS demand.
Therefore, there is currently no efficient way to offer or guarantee QoS over the Internet, other packet networks, or other best-effort networks and, even with call admission networks, there is no effective manner for dealing with missed performance levels. In general, all telecommunication links are error prone, to some level. Service providers can profit by allowing increased error rates and/or latencies and will be tempted to do so. However, users generally have no mechanism to determine when such degradations occur and no mechanism allowing them to be compensated even if they determine such degradations are occurring.
Scott Jordan and Hong Jiang survey a number of models, in which the network offers a rate schedule from which the calling party selects their preference, in xe2x80x9cConnection Establishment in High-Speed Networksxe2x80x9d, IEEE Journal on Selected Areas in Communications, vol. 13, no. 7, September 1995. Nagao Ogino presents a similar methodology, in which a number of service providers bid on the provision of communication services, in xe2x80x9cConnection Establishment Protocol Based on Mutual Selection by Users and Network Providersxe2x80x9d, ACM, 1998. In both cases though, there is no discussion or consideration of how users can ensure they will obtain the performance that they paid for or how they might be compensated if they do not obtain the performance agreed with the service provider.
Users of the existing PSTN are used to very predictable quality and reliability referred to as xe2x80x9cfour 9""sxe2x80x9d reliability. That is, successful performance of a voice quality communication can be expected in 9,999 out of 10,000 calls, once a connection is obtained. To date, such reliability cannot be obtained on packet or other networks, but the much lower cost of Internet Protocol based services and increased diversity will force the PSTN to incorporate those protocols in order to compete. Clearly, some means of addressing this problem is required.
The connectionless telecommunication networks known in the art do not offer guaranteed service levels. Further, there does not exist any mechanism for warrantying communication parameters to users over connectionless, call admission or other telecommunication networks. There is therefore a need for a method and system of providing telecommunication services over connectionless and other telecommunication networks, which improves upon the problems known in the art. This design must be provided with consideration for ease of implementation and recognize the pervasiveness of existing telecommunication infrastructures.
It is therefore an object of the present invention to provide a method and system for establishing and managing communications over telecommunication networks which obviates or mitigates at least one of the disadvantages of the prior art.
According to a first aspect of the present invention, there is provided a method of communication between at least first and second entities over a telecommunication network, where said communication is defined by a set of parameters, said method comprising the steps of:
(i) negotiating between said at least first and second entities an agreed set of values for said parameters that define the desired communication;
(ii) negotiating a warranty agreement between said at least first and second entities defining at least one of said agreed parameters to be monitored and a compensation method to be applied should said at least one monitored parameter fail to meet the corresponding one of said agreed values;
(iii) establishing said communication;
(iv) monitoring said at least one parameter of said communication; and
(v) in the event of a failure of said monitored parameter to meet said agreed value, compensating at least one of said first and second entities in accordance with said negotiated compensation method.
According to another aspect of the present invention, there is provided a telecommunications system comprising:
a first end user;
a second end user;
a telecommunications network interconnecting said first end user said second end user and having at least one transmission link and protocol;
said first end user and said telecommunication network negotiating a communication between said first end user and said second end user; and
said first end user and said telecommunication network being operable to:
(a) agree on values for a set of parameters defining a communication between said first end user and said second end user; and
(b) agree on a warranty agreement defining at least one of said set of parameters to be monitored and a compensation method to be applied should said at least one monitored parameter fail to meet the corresponding one of said agreed values.
In the present invention, communications between at least two end users are achieved over at least one telecommunications link. Preferably, the communication is defined by a set of parameters, usually including one or more network performance parameters and/or QoS parameters, which are negotiated by, or on behalf of, at least one of the end users with the one or more network service providers who will establish the communication. A successful negotiation results in an agreed set of values for the parameters and a warranty agreement with the network service provider that defines at least one of the agreed parameter values to be warranted. The warranty agreement also defines a compensation method to be applied should a measured value of the parameter fail to meet the corresponding agreed value.
Once the communication is established, the agreed warranted parameter, or parameters, are monitored and, in the event of a failure of a monitored parameter to meet the agreed values, the compensation method is invoked and at least one user or other entity involved in the communication is compensated in accordance with the compensation method. If multiple users are involved in a communication, the compensation can be divided amongst them according to an agreed scheme. Similarly, if multiple network service providers are involved in the communication with the users and/or in establishing point to point links in the communication, the network service provider who fails to meet agreed parameters can compensate other network service providers and/or users, as appropriate. Compensation can be achieved in a variety of manners, including by monetary means, including reduced billings, refunds and/or penalty payments, or by the provision of free or reduced rate communications, either for the present communication or for a future communication.
The telecommunication network can be a call admission network, a connectionless network, a virtual connection network or any combination of these networks. In the case of call admission networks, the negotiation of values for communication parameters can be trivial, but a warranty can still be agreed and provided.